1. Field of the Invention
The present invention relates to a system for providing transparent access to different types of communication networks that may be incompatible with each other and some of which may be incompatible; with the equipment used by the calling party or the called party, least cost routing in such a system, maintaining quality of communication in such a system, prioritizing the routing of such communications, evaluating different communication access locations to determine where to send a communication, synchronizing communications, blocking incoming communications while waiting for the synchronizing to be completed, and minimizing the cost of communications using such a system. This system also monitors and records the services used on each of the unrelated service providers. This information is then utilized for billing purposes and for paying the service providers.
The present invention also relates to the managing of communications through the IP network. In particular, the present invention relates to the optimization of the routing of multimedia communications among user devices across convergent networks.
2. Description of the Related Art
Presently when communication services are offered on a global basis, communications are established through the equipment of a plurality of service providers located in various countries. This communication is dominated by large carriers which have formed the global network through reciprocal agreements. Smaller competing carriers, who may offer the same service at lower prices, currently do not have reciprocal agreements between them.
The invention provides these smaller competing carriers with access to each other without the use of the large carriers. Such access provides the calling party (e.g., a subscriber of the smaller competing carrier) with the option of obtaining optimum service at lower prices while ensuring that the appropriate service providers get paid. The calling party can now have cheaper access to different types of telecommunication networks that the party may not have access to under the current large carrier system. It may be cheaper or preferred for the calling party to use smaller carriers to communicate with another location by routing the communication over a digital data network rather than an analog voice network, or by routing the communication over a paging network rather than a cellular network or a combination of networks.
With computing devices rapidly evolving to include sophisticated communicating functions, consumers or end users are becoming more and more exposed to the possibilities of communicating with each other via audio and visual media. Voice calls no longer suffice as a means of communication. This phenomenon is what some may call telecommunications media convergence which transcends traditional telecom industries such as fixed, mobile, and IP service providers. Convergence is the combination of different media into one operating platform. Thus, a convergent network (as used herein) is a network comprising various protocol-specific networks such as circuit-switched, mobile, and IP networks which are interconnected with each other. It is the merger of telecom, data processing and imaging technologies. This convergence is shepherding in a new era of multimedia communication, wherein voice, data, images and video are merged and become part and parcel of any telecommunications services demanded by the end users.
In this convergent world, the network operators must be capable of routing high quality multimedia contents between fixed or mobile devices such as, for example, smart phones, laptops, iPads, desktops, and audio-video equipment. To provide a quality user experience, network operators need to ensure their networks have the requisite or appropriate transmission characteristics such as bandwidth, latency, and jitter in the case of an IP network for the transmission of multimedia content. However, traditional network operators' ability to choose routes are confined to their own networks and typically do not have control over communications that transcend across multiple networks. Moreover, the transmission of multimedia content, especially broadcast quality high definition video, requires the networks to transport the content with high fidelity, i.e. with little or no loss of data. This is a difficult task in a world where the IP networks dominate and offer the least cost alternative for content transmission, but which are notorious for latency and packet loss. When the multimedia communications occur over the disparate networks of different technologies and protocols, the management of high quality multimedia communications can become insurmountable or very expensive.
Accordingly, there is a need for a cost-effective system that can manage and selectively route multimedia communications among multiple parties, transparent and seamless to the users, through one or more service providers or network operators based on user requirements. Such requirements may be based on, for example, whether the content comprises high definition video or merely voice data coupled with low fidelity video or based on the hardware and/or client software characteristics of the access devices. In the case of internet service providers, and as explained below, the system can perform practical quality test measurements of each route available or offered by the internet service providers for routing subscriber communications traffic.
Some insight into the workings of the Internet is in order. It is widely known that the Internet is a worldwide network of interconnected networks. Each individual host connected to the internet has an IP address. To send a data packet from one host to another, the data packet must be routed through the Internet. To accomplish this, each host includes a routing table the host uses to determine which physical interface address to use for sending the data. When a host receives a data packet, the data packet is either intended for that host or intended for another host. When the latter occurs, the host retransmits the packet using its own route table. Route tables are based on static rules or dynamic rules via routing protocols. Accordingly, the quality of the route depends on the quality of each host that the packet passes through and the network elements that connect the hosts. It would be useful to know the quality level of each particular host along a route so that packets requiring a higher quality could be routed using hosts having a high quality measurement score.
Some individual quality indicators such as, for example, latency, availability, packet loss may be determined for certain routes on the Internet. However, depending on the type of multimedia content to be delivered, the best route for one application (e.g. near real-time broadcast) may be the route with the lowest latency characteristics, while the best route for another application (e.g. high definition video) may require the least packet loss characteristic.